| |
| % lame [options] inputfile [outputfile] |
| |
| For more options, just type: |
| % lame --help |
| |
| |
| ======================================================================= |
| Constant Bitrate Examples: |
| ======================================================================= |
| fixed bit rate jstereo 128 kbps encoding: |
| % lame sample.wav sample.mp3 |
| |
| fixed bit rate jstereo 128 kbps encoding, higher quality: (recommended) |
| % lame -h sample.wav sample.mp3 |
| |
| Fast encode, low quality (no noise shaping) |
| % lame -f sample.wav sample.mp3 |
| |
| ======================================================================= |
| Variable Bitrate Examples: |
| ======================================================================= |
| LAME has two types of variable bitrate: ABR and VBR. |
| |
| ABR is the type of variable bitrate encoding usually found in other |
| MP3 encoders, Vorbis and AAC. The number of bits is determined by |
| some metric (like perceptual entropy, or just the number of bits |
| needed for a certain set of encoding tables), and it is not based on |
| computing the actual encoding/quantization error. ABR should always |
| give results equal or better than CBR: |
| |
| ABR: (--abr <x> means encode with an average bitrate of around x kbps) |
| lame -h --abr 128 sample.wav sample.mp3 |
| |
| |
| VBR is a true variable bitrate mode which bases the number of bits for |
| each frame on the measured quantization error relative to the |
| estimated allowed masking. There are 10 compression levels defined, |
| ranging from 0=lowest compression to 9 highest compression. The resulting |
| filesizes depend on the input material. On typical music you can expect |
| -V5 resulting in files averaging 132 kbps, -V2 averaging 200 kbps. |
| |
| Variable Bitrate (VBR): (use -V n to adjust quality/filesize) |
| % lame -V2 sample.wav sample.mp3 |
| |
| |
| |
| ======================================================================= |
| LOW BITRATES |
| ======================================================================= |
| At lower bitrates, (like 24 kbps per channel), it is recommended that |
| you use a 16 kHz sampling rate combined with lowpass filtering. LAME, |
| as well as commercial encoders (FhG, Xing) will do this automatically. |
| However, if you feel there is too much (or not enough) lowpass |
| filtering, you may need to try different values of the lowpass cutoff |
| and passband width (--resample, --lowpass and --lowpass-width options). |
| |
| |
| ======================================================================= |
| STREAMING EXAMPLES |
| ======================================================================= |
| |
| % cat inputfile | lame [options] - - > output |
| |
| |
| |
| |
| ======================================================================= |
| Scripts are included (in the 'misc' subdirectory) |
| to run lame on multiple files: |
| |
| bach script: mlame Run "mlame -?" for instructions. |
| sh script: auenc Run auenc for instructions |
| sh script: mugeco.sh |
| |
| Pearl script which will re-encode mp3 files and preserve id3 tags: |
| lameid3.pl |
| |
| Windows scripts: |
| lame4dos.bat |
| Lame.vbs (and an HTML frontend: LameGUI.html) |
| |
| |
| ======================================================================= |
| options guide: |
| ======================================================================= |
| These options are explained in detail below. |
| |
| |
| Quality related: |
| |
| -m m/s/j/f/a mode selection |
| -q n Internal algorithm quality setting 0..9. |
| 0 = slowest algorithms, but potentially highest quality |
| 9 = faster algorithms, very poor quality |
| -h same as -q2 |
| -f same as -q7 |
| |
| |
| Constant Bit Rate (CBR) |
| -b n set bitrate (8, 16, 24, ..., 320) |
| --freeformat produce a free format bitstream. User must also specify |
| a bitrate with -b, between 8 and 640 kbps. |
| |
| Variable Bit Rate (VBR) |
| -v VBR |
| --vbr-old use old variable bitrate (VBR) routine |
| --vbr-new use new variable bitrate (VBR) routine (default) |
| -V n VBR quality setting (0=highest quality, 9=lowest) |
| -b n specify a minimum allowed bitrate (8,16,24,...,320) |
| -B n specify a maximum allowed bitrate (8,16,24,...,320) |
| -F strictly enforce minimum bitrate |
| -t disable VBR informational tag |
| --nohist disable display of VBR bitrate histogram |
| |
| --abr n specify average bitrate desired |
| |
| |
| Operational: |
| |
| -r assume input file is raw PCM |
| -s n input sampling frequency in kHz (for raw PCM input files) |
| --resample n output sampling frequency |
| --mp3input input file is an MP3 file. decode using mpglib/mpg123 |
| --ogginput input file is an Ogg Vorbis file. decode using libvorbis |
| -x swap bytes of input file |
| --scale <arg> multiply PCM input by <arg> |
| --scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg> |
| --scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg> |
| -a downmix stereo input file to mono .mp3 |
| -e n/5/c de-emphasis |
| -p add CRC error protection |
| -c mark the encoded file as copyrighted |
| -o mark the encoded file as a copy |
| -S don't print progress report, VBR histogram |
| --strictly-enforce-ISO comply as much as possible to ISO MPEG spec |
| --replaygain-fast compute RG fast but slightly inaccurately (default) |
| --replaygain-accurate compute RG more accurately and find the peak sample |
| --noreplaygain disable ReplayGain analysis |
| --clipdetect enable --replaygain-accurate and print a message whether |
| clipping occurs and how far the waveform is from full scale |
| |
| --decode assume input file is an mp3 file, and decode to wav. |
| -t disable writing of WAV header when using --decode |
| (decode to raw pcm, native endian format (use -x to swap)) |
| |
| |
| |
| ID3 tagging: |
| |
| --tt <title> audio/song title (max 30 chars for version 1 tag) |
| --ta <artist> audio/song artist (max 30 chars for version 1 tag) |
| --tl <album> audio/song album (max 30 chars for version 1 tag) |
| --ty <year> audio/song year of issue (1 to 9999) |
| --tc <comment> user-defined text (max 30 chars for v1 tag, 28 for v1.1) |
| --tn <track> audio/song track number (1 to 255, creates v1.1 tag) |
| --tg <genre> audio/song genre (name or number in list) |
| --add-id3v2 force addition of version 2 tag |
| --id3v1-only add only a version 1 tag |
| --id3v2-only add only a version 2 tag |
| --space-id3v1 pad version 1 tag with spaces instead of nulls |
| --pad-id3v2 same as '--pad-id3v2-size 128' |
| --pad-id3v2-size <num> adds version 2 tag, pad with extra <num> bytes |
| --genre-list print alphabetically sorted ID3 genre list and exit |
| |
| Note: A version 2 tag will NOT be added unless one of the input fields |
| won't fit in a version 1 tag (e.g. the title string is longer than 30 |
| characters), or the '--add-id3v2' or '--id3v2-only' options are used, |
| or output is redirected to stdout. |
| |
| Windows and OS/2-specific options: |
| --priority <type> sets the process priority |
| |
| |
| options not yet described: |
| --nores disable bit reservoir |
| --disptime |
| |
| --lowpass |
| --lowpass-width |
| --highpass |
| --highpass-width |
| |
| |
| |
| |
| |
| ======================================================================= |
| Detailed description of all options in alphabetical order |
| ======================================================================= |
| |
| |
| ======================================================================= |
| downmix |
| ======================================================================= |
| -a |
| |
| mix the stereo input file to mono and encode as mono. |
| |
| This option is only needed in the case of raw PCM stereo input |
| (because LAME cannot determine the number of channels in the input file). |
| To encode a stereo PCM input file as mono, use "lame -m s -a" |
| |
| For WAV and AIFF input files, using "-m m" will always produce a |
| mono .mp3 file from both mono and stereo input. |
| |
| |
| ======================================================================= |
| average bitrate encoding (aka Safe VBR) |
| ======================================================================= |
| --abr n |
| |
| turns on encoding with a targeted average bitrate of n kbps, allowing |
| to use frames of different sizes. The allowed range of n is 8...320 |
| kbps, you can use any integer value within that range. |
| |
| |
| |
| |
| |
| ======================================================================= |
| bitrate |
| ======================================================================= |
| -b n |
| |
| For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz) |
| n = 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 |
| |
| For MPEG-2 (sampling frequencies of 16, 22.05 and 24 kHz) |
| n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 |
| |
| For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz) |
| n = 8, 16, 24, 32, 40, 48, 56, 64 |
| |
| |
| The bitrate to be used. Default is 128 kbps MPEG1, 80 kbps MPEG2. |
| |
| When used with variable bitrate encodings (VBR), -b specifies the |
| minimum bitrate to use. This is useful only if you need to circumvent |
| a buggy hardware device with strange bitrate constrains. |
| |
| |
| ======================================================================= |
| max bitrate |
| ======================================================================= |
| -B n |
| |
| see also option "-b" for allowed bitrates. |
| |
| Maximum allowed bitrate when using VBR/ABR. |
| |
| Using -B is NOT RECOMMENDED. A 128 kbps CBR bitstream, because of the |
| bit reservoir, can actually have frames which use as many bits as a |
| 320 kbps frame. ABR/VBR modes minimize the use of the bit reservoir, and |
| thus need to allow 320 kbps frames to get the same flexability as CBR |
| streams. This is useful only if you need to circumvent a buggy hardware |
| device with strange bitrate constrains. |
| |
| |
| |
| |
| ======================================================================= |
| copyright |
| ======================================================================= |
| -c |
| |
| mark the encoded file as copyrighted |
| |
| |
| |
| ======================================================================= |
| clipping detection |
| ======================================================================= |
| --clipdetect |
| |
| Enable --replaygain-accurate and print a message whether clipping |
| occurs and how far in dB the waveform is from full scale. |
| |
| This option is not usable if the MP3 decoder was _explicitly_ disabled |
| in the build of LAME. |
| |
| See also: --replaygain-accurate |
| |
| |
| |
| ======================================================================= |
| mpglib decode capability |
| ======================================================================= |
| --decode |
| |
| This just uses LAME's mpg123/mpglib interface to decode an MP3 file to |
| a wav file. The input file can be any input type supported by |
| encoding, including .mp3 (layers 1, 2 and 3) and .ogg. |
| |
| If -t is used (disable wav header), LAME will output |
| raw pcm in native endian format (use -x to swap bytes). |
| |
| This option is not usable if the MP3 decoder was _explicitly_ disabled |
| in the build of LAME. |
| |
| |
| ======================================================================= |
| de-emphasis |
| ======================================================================= |
| -e n/5/c |
| |
| n = (none, default) |
| 5 = 0/15 microseconds |
| c = citt j.17 |
| |
| All this does is set a flag in the bitstream. If you have a PCM |
| input file where one of the above types of (obsolete) emphasis has |
| been applied, you can set this flag in LAME. Then the mp3 decoder |
| should de-emphasize the output during playback, although most |
| decoders ignore this flag. |
| |
| A better solution would be to apply the de-emphasis with a standalone |
| utility before encoding, and then encode without -e. |
| |
| |
| |
| ======================================================================= |
| fast mode |
| ======================================================================= |
| -f |
| |
| Same as -q 7. |
| |
| NOT RECOMMENDED. Use when encoding speed is critical and encoding |
| quality does not matter. Disable noise shaping. Psycho acoustics are |
| used only for bit allocation and pre-echo detection. |
| |
| ======================================================================= |
| strictly enforce VBR minimum bitrate |
| ======================================================================= |
| -F |
| |
| strictly enforce VBR minimum bitrate. With out this optioni, the minimum |
| bitrate will be ignored for passages of analog silence. |
| |
| |
| |
| ======================================================================= |
| free format bitstreams |
| ======================================================================= |
| --freeformat |
| |
| LAME will produce a fixed bitrate, free format bitstream. |
| User must specify the desired bitrate in kbps, which can |
| be any integer between 8 and 640. |
| |
| Not supported by most decoders. Complient decoders (of which there |
| are few) are only required to support up to 320 kbps. |
| |
| Decoders which can handle free format: |
| |
| supports up to |
| MAD 640 kbps |
| "lame --decode" 550 kbps |
| Freeamp: 440 kbps |
| l3dec: 310 kbps |
| |
| |
| |
| |
| |
| ======================================================================= |
| high quality |
| ======================================================================= |
| -h |
| |
| use some quality improvements. The same as -q 2. |
| |
| |
| |
| ======================================================================= |
| Modes: |
| ======================================================================= |
| |
| -m m mono |
| -m s stereo |
| -m j joint stereo |
| -m f forced mid/side stereo |
| -m d dual (independent) channels |
| -m i intensity stereo |
| -m a auto |
| |
| MONO is the default mode for mono input files. If "-m m" is specified |
| for a stereo input file, the two channels will be averaged into a mono |
| signal. |
| |
| STEREO |
| |
| JOINT STEREO is the default mode for stereo files with fixed bitrates of |
| 128 kbps or less. At higher fixed bitrates, the default is stereo. |
| For VBR encoding, jstereo is the default for VBR_q >4, and stereo |
| is the default for VBR_q <=4. You can override all of these defaults |
| by specifing the mode on the command line. |
| |
| jstereo means the encoder can use (on a frame by frame bases) either |
| regular stereo (just encode left and right channels independently) |
| or mid/side stereo. In mid/side stereo, the mid (L+R) and side (L-R) |
| channels are encoded, and more bits are allocated to the mid channel |
| than the side channel. This will effectively increase the bandwidth |
| if the signal does not have too much stereo separation. |
| |
| Mid/side stereo is basically a trick to increase bandwidth. At 128 kbps, |
| it is clearly worth while. At higher bitrates it is less useful. |
| |
| For truly mono content, use -m m, which will automatically down |
| sample your input file to mono. This will produce 30% better results |
| over -m j. |
| |
| Using mid/side stereo inappropriately can result in audible |
| compression artifacts. To much switching between mid/side and regular |
| stereo can also sound bad. To determine when to switch to mid/side |
| stereo, LAME uses a much more sophisticated algorithm than that |
| described in the ISO documentation. |
| |
| FORCED MID/SIDE STEREO forces all frames to be encoded mid/side stereo. It |
| should only be used if you are sure every frame of the input file |
| has very little stereo seperation. |
| |
| DUAL CHANNELS Not supported. |
| |
| INTENSITY STEREO |
| |
| AUTO |
| |
| Auto select should select (if input is stereo) |
| 8 kbps Mono |
| 16- 96 kbps Intensity Stereo (if available, otherwise Joint Stereo) |
| 112-128 kbps Joint Stereo -mj |
| 160-192 kbps -mj with variable mid/side threshold |
| 224-320 kbps Independent Stereo -ms |
| |
| |
| |
| ======================================================================= |
| MP3 input file |
| ======================================================================= |
| --mp3input |
| |
| Assume the input file is a MP3 file. LAME will decode the input file |
| before re-encoding it. Since MP3 is a lossy format, this is |
| not recommended in general. But it is useful for creating low bitrate |
| mp3s from high bitrate mp3s. If the filename ends in ".mp3" LAME will assume |
| it is an MP3. For stdin or MP3 files which dont end in .mp3 you need |
| to use this switch. |
| |
| |
| ======================================================================= |
| disable historgram display |
| ======================================================================= |
| --nohist |
| |
| By default, LAME will display a bitrate histogram while producing |
| VBR mp3 files. This will disable that feature. |
| |
| |
| ======================================================================= |
| disable ReplayGain analysis |
| ======================================================================= |
| --noreplaygain |
| |
| By default ReplayGain analysis is enabled. This switch disables it. |
| |
| See also: --replaygain-accurate, --replaygain-fast |
| |
| |
| ======================================================================= |
| non-original |
| ======================================================================= |
| -o |
| |
| mark the encoded file as a copy |
| |
| |
| |
| ======================================================================= |
| CRC error protection |
| ======================================================================= |
| -p |
| |
| turn on CRC error protection. |
| Yes this really does work correctly in LAME. However, it takes |
| 16 bits per frame that would otherwise be used for encoding. |
| |
| |
| ======================================================================= |
| algorithm quality selection |
| ======================================================================= |
| -q n |
| |
| Bitrate is of course the main influence on quality. The higher the |
| bitrate, the higher the quality. But for a given bitrate, |
| we have a choice of algorithms to determine the best |
| scalefactors and huffman encoding (noise shaping). |
| |
| -q 0: use slowest & best possible version of all algorithms. |
| |
| -q 2: recommended. Same as -h. -q 0 and -q 1 are slow and may not produce |
| significantly higher quality. |
| |
| -q 5: default value. Good speed, reasonable quality |
| |
| -q 7: same as -f. Very fast, ok quality. (psycho acoustics are |
| used for pre-echo & M/S, but no noise shaping is done. |
| |
| -q 9: disables almost all algorithms including psy-model. poor quality. |
| |
| |
| |
| ======================================================================= |
| input file is raw pcm |
| ======================================================================= |
| -r |
| |
| Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo |
| must be specified on the command line. Without -r, LAME will perform |
| several fseek()'s on the input file looking for WAV and AIFF headers. |
| |
| Not supported if LAME is compiled to use LIBSNDFILE. |
| |
| |
| |
| ======================================================================= |
| slightly more accurate ReplayGain analysis and finding the peak sample |
| ======================================================================= |
| --replaygain-accurate |
| |
| Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded |
| data stream. Find the peak sample of the decoded data stream and store |
| it in the file. |
| |
| |
| ReplayGain analysis does _not_ affect the content of a compressed data |
| stream itself, it is a value stored in the header of a sound file. |
| Information on the purpose of ReplayGain and the algorithms used is |
| available from http://www.replaygain.org/ |
| |
| By default, LAME performs ReplayGain analysis on the input data (after |
| the user-specified volume scaling). This behaviour might give slightly |
| inaccurate results because the data on the output of a lossy |
| compression/decompression sequence differs from the initial input data. |
| When --replaygain-accurate is specified the mp3 stream gets decoded on |
| the fly and the analysis is performed on the decoded data stream. |
| Although theoretically this method gives more accurate results, it has |
| several disadvantages: |
| * tests have shown that the difference between the ReplayGain values |
| computed on the input data and decoded data is usually no greater |
| than 0.5dB, although the minimum volume difference the human ear |
| can perceive is about 1.0dB |
| * decoding on the fly significantly slows down the encoding process |
| The apparent advantage is that: |
| * with --replaygain-accurate the peak sample is determined and |
| stored in the file. The knowledge of the peak sample can be useful |
| to decoders (players) to prevent a negative effect called 'clipping' |
| that introduces distortion into sound. |
| |
| |
| Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag. |
| The analysis is performed with the reference volume equal to 89dB. |
| Note: the reference volume has been changed from 83dB on transition |
| from version 3.95 to 3.95.1. |
| |
| This option is not usable if the MP3 decoder was _explicitly_ disabled |
| in the build of LAME. (Note: if LAME is compiled without the MP3 decoder, |
| ReplayGain analysis is performed on the input data after user-specified |
| volume scaling). |
| |
| See also: --replaygain-fast, --noreplaygain, --clipdetect |
| |
| |
| ======================================================================= |
| fast ReplayGain analysis |
| ======================================================================= |
| --replaygain-fast |
| |
| Compute "Radio" ReplayGain of the input data stream after user-specified |
| volume scaling and/or resampling. |
| |
| ReplayGain analysis does _not_ affect the content of a compressed data |
| stream itself, it is a value stored in the header of a sound file. |
| Information on the purpose of ReplayGain and the algorithms used is |
| available from http://www.replaygain.org/ |
| |
| Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag. |
| The analysis is performed with the reference volume equal to 89dB. |
| Note: the reference volume has been changed from 83dB on transition |
| from version 3.95 to 3.95.1. |
| |
| This switch is enabled by default. |
| |
| See also: --replaygain-accurate, --noreplaygain |
| |
| |
| |
| ======================================================================= |
| output sampling frequency in kHz |
| ======================================================================= |
| --resample n |
| |
| where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48 |
| |
| Output sampling frequency. Resample the input if necessary. |
| |
| If not specified, LAME may sometimes resample automatically |
| when faced with extreme compression conditions (like encoding |
| a 44.1 kHz input file at 32 kbps). To disable this automatic |
| resampling, you have to use --resamle to set the output samplerate |
| equal to the inptu samplerate. In that case, LAME will not |
| perform any extra computations. |
| |
| |
| |
| ======================================================================= |
| sampling frequency in kHz |
| ======================================================================= |
| -s n |
| |
| where n = sampling rate in kHz. |
| |
| Required for raw PCM input files. Otherwise it will be determined |
| from the header information in the input file. |
| |
| LAME will automatically resample the input file to one of the |
| supported MP3 samplerates if necessary. |
| |
| |
| ======================================================================= |
| silent operation |
| ======================================================================= |
| -S |
| |
| don't print progress report |
| |
| ======================================================================= |
| scale |
| ======================================================================= |
| --scale <arg> |
| |
| Scales input by <arg>. This just multiplies the PCM data |
| (after it has been converted to floating point) by <arg>. |
| |
| <arg> > 1: increase volume |
| <arg> = 1: no effect |
| <arg> < 1: reduce volume |
| |
| Use with care, since most MP3 decoders will truncate data |
| which decodes to values greater than 32768. |
| |
| |
| ======================================================================= |
| strict ISO complience |
| ======================================================================= |
| --strictly-enforce-ISO |
| |
| With this option, LAME will enforce the 7680 bit limitation on |
| total frame size. This results in many wasted bits for |
| high bitrate encodings. |
| |
| |
| ======================================================================= |
| disable VBR tag |
| ======================================================================= |
| -t |
| |
| Disable writing of the VBR Tag (only valid if -v flag is |
| specified) This tag in embedded in frame 0 of the MP3 file. It lets |
| VBR aware players correctly seek and compute playing times of VBR |
| files. |
| |
| When '--decode' is specified (decode mp3 to wav), this flag will |
| disable writing the WAV header. The output will be raw pcm, |
| native endian format. Use -x to swap bytes. |
| |
| |
| |
| ======================================================================= |
| variable bit rate (VBR) |
| ======================================================================= |
| -v |
| |
| Turn on VBR. There are several ways you can use VBR. I personally |
| like using VBR to get files slightly bigger than 128 kbps files, where |
| the extra bits are used for the occasional difficult-to-encode frame. |
| For this, try specifying a minimum bitrate to use with VBR: |
| |
| lame -v -b 112 input.wav output.mp3 |
| |
| If the file is too big, use -V n, where n = 0...9 |
| |
| lame -v -V n -b 112 input.wav output.mp3 |
| |
| |
| If you want to use VBR to get the maximum compression possible, |
| and for this, you can try: |
| |
| lame -v input.wav output.mp3 |
| lame -v -V n input.wav output.mp3 (to vary quality/filesize) |
| |
| |
| |
| |
| |
| |
| ======================================================================= |
| VBR quality setting |
| ======================================================================= |
| -V n |
| |
| n = 0...9. Specifies the value of VBR_q. |
| default = 4, highest quality = 0, smallest files = 9 |
| |
| Using -V 6 or higher (lower quality) is NOT RECOMMENDED. |
| ABR will produce better results. |
| |
| |
| How is VBR_q used? |
| |
| The value of VBR_q influences two basic parameters of LAME's psycho |
| acoustics: |
| a) the absolute threshold of hearing |
| b) the sample to noise ratio |
| The lower the VBR_q value the lower the injected quantization noise |
| will be. |
| |
| *NOTE* No psy-model is perfect, so there can often be distortion which |
| is audible even though the psy-model claims it is not! Thus using a |
| small minimum bitrate can result in some aggressive compression and |
| audible distortion even with -V 0. Thus using -V 0 does not sound |
| better than a fixed 256 kbps encoding. For example: suppose in the 1 kHz |
| frequency band the psy-model claims 20 dB of distortion will not be |
| detectable by the human ear, so LAME VBR-0 will compress that |
| frequency band as much as possible and introduce at most 20 dB of |
| distortion. Using a fixed 256 kbps framesize, LAME could end up |
| introducing only 2 dB of distortion. If the psy-model was correct, |
| they will both sound the same. If the psy-model was wrong, the VBR-0 |
| result can sound worse. |
| |
| |
| ======================================================================= |
| swapbytes |
| ======================================================================= |
| -x |
| |
| swap bytes in the input file (and output file when using --decode). |
| For sorting out little endian/big endian type problems. If your |
| encodings sound like static, try this first. |
| |
| ======================================================================= |
| Window and OS/2 process priority control |
| ======================================================================= |
| --priority <type> |
| |
| (Windows and OS/2 only) |
| |
| Sets the process priority for LAME while running under IBM OS/2. |
| This can be very useful to avoid the system becoming slow and/or |
| unresponsive. By setting LAME to run in a lower priority, you leave |
| more time for the system to update basic processing (drawing windows, |
| polling keyboard/mouse, etc). The impact in LAME's performance is |
| minimal if you use priority 0 to 2. |
| |
| The valid parameters are: |
| |
| 0 = Low priority (IDLE, delta = 0) |
| 1 = Medium priority (IDLE, delta = +31) |
| 2 = Regular priority (REGULAR, delta = -31) |
| 3 = High priority (REGULAR, delta = 0) |
| 4 = Maximum priority (REGULAR, delta = +31) |
| |
| Note that if you call '--priority' without a parameter, then |
| priority 0 will be assumed. |
| |
| |
| |
| |
